search for: phones1

...yes ; Typically set to NO if behind NAT allow=all =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- extensions.conf [general] static=yes writeprotect=no [globals] TRUNK=Gradwell TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) PHONES1=SIP/2201 [flat] include => home include => outgoing [home] exten => 2201,1,Dial(${PHONES1},20,Ttm) exten => 2201,2,Macro(vmessage,${PHONES1VM}) exten => 2201,3,Hangup [outgoing] ignorepat => 9 ignorepat => 8 exten => _9.,1,Dial(SIP/${EXTEN:1}@Talklite) exten => _8.,1,...

...aw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.0.203:29422 Looking for Voicemail in internal (domain 10.2.0.2) list_route: hop: <sip:1101 at 10.2.0.203:5060;transport=udp> cworks-phones1*CLI> <--- Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bKddc0a0b8;received=10.2.0.203 From: "Jim Felderman" <sip:1101 at 10.2.0.2>;tag=001d45b61d4906943a6bb290-9435a462 To: <sip:Voicemail at 10.2.0.2> Ca...

OK, this may seem to be an obvious question but where do I find the reference docs? I'm getting this error message: Timeout, but no rule 't' in context 'home' about this line: exten => 2201,1,Dial(${PHONES1},20,Ttm) I know the problem is with the 't' but I don't know what the parameters mean. I looking for a man page basically. -- Linux Home Automation Neil Cherry ncherry@comcast.net http://home.comcast.net/~ncherry/ (Text only) http://linuxha.sourceforge.ne...

I'm able to receive but not make calls with zaptel using an X101P connecting to Asterisk with an Xlite client. My client has context = flat in sip.conf and extensions number 8919 In extensions.conf I've got: [home] ; Line 1 ; exten => 8919,1,Dial(${PHONES1},20,Ttm) exten => 8919,2,Macro(vmessage,${PHONES1VM}) exten => 8919,3,Hangup [outgoing] exten => _9.,1,Dial(Zap/1/$EXTEN:1) [flat] include => home include => outgoing zapata.conf contains the following - I have 2 x101p cards installed [channels] language=en group=1 context=from-...

...aw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.0.203:24394 Looking for Voicemail in internal (domain 10.2.0.2) list_route: hop: <sip:1101 at 10.2.0.203:5060;transport=udp> cworks-phones1*CLI> <--- Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203 From: "Jim Felderman" <sip:1101 at 10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b To: <sip:Voicemail at 10.2.0.2> Ca...

Actually doing both sounds good to me. Can you explain further about ringing them all at once? Here is how I tried to make mine work and failed... {global} PHONES0=SIP/2000 PHONES1=SIP/2001 [local] exten => 6001,1,Dial(${PHONES0&PHONES1),20,trf) When I dial 6001 I see my debugger tell me that I am using the wrong syntax. Do you know the correct syntax for ringing them all at once? I will check out the queue system you referenced. Thanks! Wiley -----Original M...

Can you also post information such as: Type of phone (model Number would be idela) How is it conencted, SIP, ZAP, IAX, Channel Bank. Corresponding config files would also help. Help us help you. >>-----Original Message----- >>From: asterisk-users-bounces@lists.digium.com >>[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of >>Paul A Brown >>Sent:

...nd host=192.168.2.29 context=home secret=******** callerid="OFFICE PHONE #2" <7890> mailbox=7890 dtmfmode=rfc2833 nat=0 AND HERE IS MY EXTENSIONS.CONF FILE [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp PHONES1=SIP/7890 ; Phone 1 Def PHONES1VM=7890 ; Phone 1 VM Def FWDUSERID1=691657 MYNAME1=My name MYPHONE1=691657 TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [fwd-forced-fwd1] ; Check to see if the called number starts with a "7" and ; if so, set the call parameters an...

...t; - <users name> - <extension number> ;------------------------------------------------ [2201] type=friend host=192.192.192.220 context=home secret=xxxxxx callerid="Paul" <2201> mailbox=2201 dtmfmode=rfc2833 nat=no EXTENSIONS.CONF writeprotect=no [globals] PHONES1=SIP/2201 PHONES1VM=2201 PHONES2=SIP/2202 PHONES2VM=2202 CONSOLE=Console/dsp ; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest ; IAXtel username/password ;IAXINFO=myuser:mypass TRUNK=Zap/g2 ; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1...

Hello All, I have some Polycom IP 500 phones that I would like to have configured for direct dialing to our voice mail system. So far I have been unable to get the hard button labeled Voice Mail to connect to Asterisk without first passing through the message center prompts. I have followed all the Admin Guide instructions regarding the phones .cfg files and using

...t; - <users name> - <extension number> ;------------------------------------------------ [2201] type=friend host=192.192.192.220 context=home secret=xxxxxx callerid="Paul" <2201> mailbox=2201 dtmfmode=rfc2833 nat=no EXTENSIONS.CONF writeprotect=no [globals] PHONES1=SIP/2201 PHONES1VM=2201 PHONES2=SIP/2202 PHONES2VM=2202 CONSOLE=Console/dsp ; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest ; IAXtel username/password ;IAXINFO=myuser:mypass TRUNK=Zap/g2 ; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1...

...icepulse or others I know that I can simply have one registration statement along with an inbound context, then in extension.conf map the outbound context. from iax.conf: register => in-xxxxxxxx:xxxxxxxx@gw5.voicepulse.com from extensions.conf [voicepulse-in] exten => 212xxxxxxx,1,Dial(${PHONES1}&${PHONES2},30) exten => 212xxxxxxx,2,Voicemail2(u${PHONES1VM}) exten => 212xxxxxxx,3,Hangup I know this way I only have to register once, but can receive calls on several inbound DID numbers without any problem, provided they are all mapped similar to what I have above within extensio...

> Timeout, but no rule 't' in context 'home' > > about this line: > > exten => 2201,1,Dial(${PHONES1},20,Ttm) > > I know the problem is with the 't' but I don't know > what the parameters mean. I looking for a man page basically. The problem isn't related to the "t" in the Dial() command, which enables call transfer, but to a missing "t" (timeout) ext...

...hone [sip-forced] exten => _3.,1,SetCallerID(${SIPPHONEUSERID}) exten => _3.,2,SetCIDName(${MYNAME}) exten => _3.,3,Dial(SIP/${EXTEN:1}@proxy01.sipphone.com) exten => _3.,4,Playback(invalid) exten => _3.,5,Hangup [from-sipphone] exten => ${SIPPHONEUSERID},1,Dial(${PHONES1},30,Ttm) exten => ${SIPPHONEUSERID},2,Voicemail2(u${PHONES1VM}) exten => ${SIPPHONEUSERID},3,Hangup -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050103/8e48fa14/attachment.htm

Hi My head hurts... Can anyone help out here, my remote IAX can see my local IAX and visa versa, conversation starts, I can dial my remote (POTS) landline number, remote end answers, trys to route to local iax2, I see it start the conversation here, the extension (SIP) rings once and then it dies... Both ends are defined with accept IPADDRESS to keep it in the family and simple.. Debug info

Trying to get SIP realtime working here... I'm connected to the database... *CLI> realtime mysql status Connected to vox180internal@db1.ipt.XXX.com, port 3306 with username voxadmin for 6 seconds. I can get information for the extension in question... *CLI> realtime load sipusers name 2944093 Column Name Column Value

...; s,3,GotoIf(${DB_EXISTS(blacklist/${CALLERID(number)})}?custom-blacklisted,s,1) exten => s,4,Set(DB(CALLTRACE/lastcaller)=${CALLERID(number)}) exten => s,5,AGI(MisterHouse.agi,"CallerID") exten => s,6,Answer exten => s,7,Playtones(ring) exten => s,8,Dial(${PHONES0}&${PHONES1}&${PHONES2}&${PHONES7}&${PHONES11},20,tr) exten => s,9,Goto(s-${DIALSTATUS},1) ; if no fax, branch on dialstatus exten => s-NOANSWER,1,Macro(voicemail,${PHONES0VM}) exten => s-NOANSWER,2,Hangup() exten => s-BUSY,1,Macro(voicemail,${PHONES0VM}) exten => s-BUSY,2,Hangup() e...

I am attempting to write in incoming context for calls. 1. If the caller id is given and it is not black listed it will Playback a greeting and then right the phone or go to voicemail under busy or unavailable conditions 2. If no caller id is given, then Privacy Manager will ask for the number. I am testing 6145551212 to see if the black list will work 3. If a caller id is given, and it is

I am configuring an asterisk server and I want to test the incoming configuration with my FXS handsets. I have the FXS lines able to call eachother and they can connect out the FXO lines. I changed the context for the FXS lines to "incoming" so that they would be able to test the setup for incoming calls. For the incoming context I have: [incoming] exten => s,1,Wait(1) exten